Apparatuses and methods to detect and obtain desired audio

ABSTRACT

A device and method to detect desired audio includes a ratio calculator. The ratio calculator calculates a ratio between a primary acoustic signal, and a reference acoustic signal. The primary acoustic signal contains desired audio and undesired audio and the reference acoustic signal contains mostly undesired audio, substantially void of undesired audio. A long-term mean value calculator is coupled to the ratio calculator. The long-term mean value calculator maintains an average of the ratio. A comparator is coupled to the ratio calculator and the long-term value calculator. The comparator compares the ratio with the average. Desired audio is detected when the ratio is greater than the average by a threshold amount.

RELATED APPLICATIONS

This application is a continuation-in-part of co-pending non-provisionalU.S. patent application Ser. No. 12/157,426 entitled “Cardioid Beam WithA Desired Null Based Acoustic Devices, Systems And Methods;” which is acontinuation of U.S. patent application Ser. No. 10/206,242 (Now U.S.Pat. No. 7,386,135), entitled “Cardioid Beam With A Desired Null BasedAcoustic Devices, Systems And Methods,” which claims priority to U.S.provisional patent application No. 60/309,462 entitled “Adaptive NoiseCancellation System,” filed on Aug. 1, 2001, U.S. patent applicationSer. No. 12/157,426 entitled “Cardioid Beam With A Desired Null BasedAcoustic Devices, Systems And Methods is hereby fully incorporated byreference. U.S. provisional patent application No. 60/309,462 entitled“Adaptive Noise Cancellation System,” is hereby fully incorporated byreference.

FIELD OF THE INVENTION

The present invention relates to the fields of acoustics and signalprocessing. More specifically, the present invention is related to audiodevices, systems and methods for sensing and/or discerning desired audioin a noisy environment, where the environmental noises are statisticallyuncorrelated to the desired audio, and located at the directions otherthan the direction of the desired audio.

BACKGROUND OF THE INVENTION

Interference from background noises is one of the main barriers toadvance acoustic applications or systems. whether it is audio acquiringmicrophone systems for communication or automatic speech recognition(ASR), hydrophone systems, sonar systems, or other acoustic systems ofthe like. The problem has found to be especially difficult with multiplebackground noise sources that are non-stationary, broadband, burstingand intermittent in a reverberant environment.

For example, in the case of ASR systems, it is increasingly desirable tointroduce ASR technology to the large number of mobile communicationdevices, such as cell phones, car phones, and PDA, recently deployed asa result of the recent rapid advances in mobile communication andrelated technologies. However, most of these devices are frequentlyoperated in a relatively noisy acoustic environment, such as on thestreet, in a car, bus, subway, train or airplane, or inside a noisymail, factory or office. The background noises of these reverberantenvironments often exhibit the earlier mentioned non-stationary,broadband, bursting and intermittent characteristics. Resultantly, newapplications utilizing speech recognition interface, whether fordictation or command-and-control, remain scarce.

To overcome these kinds of noise problems, others have resorted toclose-talk handset, headset, or ear-set devices. However, thesesolutions introduce a number of inconveniences for the users. The wiresof these additional headset/ear-set devices are often tangled with otherobjects. Wireless alternatives are more user friendly, however, theythemselves have other limitations and inconveniences, e.g., higher cost.Multi-microphone arrays may avoid some of these limitations, howeverprior art multi-microphone arrays tend to be physically large, andunsuitable for most applications.

Consequently, there is a need for a more effective solution, especiallyone that is more compact that can allow a more natural human-machineinterface that is hands-free, headset-free, and most importantlynoise-free, for certain acoustic applications, such as ASR. In addition,noise reduction and/or cancellation preferably not only increase theclarity and intelligibility of the desired audio, but thereduction/cancellation may perhaps even reduce the load of digitalcommunication networks, thereby resulting in more effective use of theircapacities.

Other applications include noise robust headphone, teleconferencingsystem, digital voice recorder and hearing aid, etc.

BRIEF DESCRIPTION OF THE DRAWINGS

The present invention will be described by way of exemplary embodiments,but not limitations, illustrated in the accompanying drawings in whichlike references denote similar elements, and in which:

FIG. 1 illustrates an overview of the present invention, in accordancewith one embodiment;

FIGS. 2a-2g illustrates beam forming of the acoustic device of FIG. 1 infurther detail, in accordance with various embodiments;

FIG. 3 illustrates supplemental logic suitable for use in conjunctionwith the acoustic device of FIG. 1 to generate the beam pattern of FIG.2g , in accordance with one embodiment;

FIG. 4a illustrates the acoustic beams generated by the various acousticdevices of FIG. 2a-2c , and FIG. 2g (when supplemented with thecircuitry of FIG. 3) in the form of a polar sensitivity plots

FIG. 4b illustrates the acoustic beams of FIG. 2d-2f in the form of apolar sensitivity plot;

FIGS. 4c-4d illustrate other possible primary signal beams in the formof polar sensitivity plots;

FIG. 5 illustrates the signal processing subsystem of FIG. 1 in furtherdetail, in accordance with one embodiment;

FIGS. 6a-6b illustrate the sampling component of FIG. 5 in furtherdetail, in accordance with one embodiment;

FIGS. 7a-7b illustrate a pre-whitening and a de-whitening componentsuitable for use as the optional signal conditioning and re-conditioningcomponents of FIG. 5 for certain acoustic applications, in accordancewith one embodiment;

FIG. 8 illustrates the signal extraction component of FIG. 5 in fartherdetail, in accordance with one embodiment;

FIG. 9 illustrates the mean amplitude computation component of FIG. 8 infurther detail, in accordance with one embodiment;

FIGS. 10a-10b illustrate the detector components of FIG. 8 in furtherdetails, in accordance with two embodiments;

FIGS. 11a-11e illustrate the echo cancellation like logic of FIG. 8, inaccordance with various embodiments;

FIG. 12 illustrates the operational logic of the inhibitor component ofFIG. 8, in accordance with one embodiment; and

FIG. 13 illustrates the signal extraction logic of FIG. 5, in accordancewith another embodiment.

SUMMARY OF THE INVENTION

Briefly, the present invention includes acoustic devices, systems andmethods.

In accordance with one aspect, an acoustic device is formed with firstand second plurality of one or more acoustic elements. The firstplurality of one or more acoustic elements are designed and arranged tofacilitate generation of a first signal that includes mostly undesiredaudio, substantially void of desired audio. The second plurality of oneor more acoustic elements are designed and arranged to facilitategeneration of a second signal that includes both the desired andundesired audio.

In one embodiment, the first one or more elements, in response to thepresence of audio, desired or undesired, output a cardioid shapedacoustic beam with a null at the originating direction of the desiredaudio. The second one or more elements output an audio beam shaped inone of a number of complementary manner to encompass or maximizing thedesired audio.

In accordance with another aspect, a signal processing subsystem isprovided to extract the desired audio, using the two signals.

In various embodiments, the signal processing subsystems may practicevarious echo cancellation like signal extraction techniques, byintroducing deterministic delays to the second signal, or practice blindsignal separation techniques.

In various echo cancellation like signal extraction embodiments,pre-whitening and de-whitening components may be provided to conditionand re-condition signals.

In one embodiment the desired audio is speech, in particular, speech tobe recognized in a noisy reverberant environment, where noise isstochastic and uncorrelated to the desired speech, such as in anautomobile or in an office.

DETAILED DESCRIPTION OF EMBODIMENTS

In the following description, various embodiments of the presentinvention will be described, in particular, ASR oriented embodiments.However, from the descriptions to follow, those skilled in the art wouldappreciate that the present invention is not limited to ASR only. Thepresent invention may be practiced in other acoustic applications,including but are not limited to communication devices, recordingdevices, hearing aids, as well as hydrophones and sonar.

For purposes of explanation, specific numbers, materials andconfigurations are set forth in order to provide a thoroughunderstanding of the present invention. However, it will be apparent tot hose skilled in the art that the present invention may be practicedwith only some of these details, and/or with other elements. In otherinstances, well-known features are omitted or simplified.

Terminology

Parts of the description will be presented in acoustic and signalprocessing terms, such as acoustic beams, impulse, response, sampling,signal conditioning, signal extractions and so forth, consistent withthe manner commonly employed by those skilled in the art to convey thesubstance of their work to others skilled in the art. As well understoodby those skilled in the art, even in software implementations of some ofthe aspects of the present invention, these quantities take the form ofelectrical, magnetic, or optical signals capable of being stored,transferred, combined, and otherwise manipulated through electricaland/or optical components of a processor and its subsystems.

Part of the descriptions will employ various abbreviations, includingbut not limited to:

-   ASR Automatic Speech Recognition-   BSS Blind Signal Separation or Blind Source Separation-   FFT Fast Fourier Transform FIR Feedback Impulse Response-   IFFT Inverse Fast Fourier Transform-   LMS Least Mean Square-   NLMS Normalized Least Mean Square    Section Headings, Order of Descriptions and Embodiments

Section headings are merely employed to improve readability, and theyare not to be construed to restrict or narrow the present invention.

Various operations will be described as multiple discrete steps in turn,in a manner that is most helpful in understanding the present invention,however, the order of description should not be construed as to implythat these operations are necessarily order dependent. In particular,these operations need not be performed in the order of presentation.

The phrase “in one embodiment” is used repeatedly. The phrase generallydoes not refer to the same embodiment, however, it may. The terms“comprising”, “having”, “including” and other constructs of the like,are synonymous, unless the context dictates otherwise.

Overview

We refer now to FIG. 1, wherein a block diagram illustrating an overviewof an acoustic system of the present invention, in accordance with oneembodiment. As illustrated, for the embodiment, acoustic system 100includes acoustic device 102 and signal processing subsystem 104, bothincorporated with the teachings of the present invention. The twoelements are coupled to each other as shown.

Acoustic device 102, in accordance with the present invention, isdesigned to respond to audio presence, desired and undesired (i.e.noise), by outputting two audio beams 103 a and 103 b, with audio beam103 a having mostly undesired audio, substantially void of desiredaudio, and audio beam 103 b having both the desired and the undesiredaudio.

The two acoustic beams (hereinafter, simply beams) are sampled by signalprocessing subsystem 104 to generate two corresponding audio signals(hereinafter, simply signals), which in turn are used by signalprocessing subsystem 104 to recover the desired audio, by removing thefirst signal corresponding to the first beam from the second signalcorresponding to the second beam.

As will be appreciated by those skilled in the art, based on thedescriptions to follow, acoustic device 102 may be formed compactlyusing as little as two acoustic elements, one each for the correspondingresponsive generation of one of the two beams. Resultantly, the presentinvention is able to provide a more compact and user-friendly humaninterface for acoustic applications that have to address the issue ofrecovering desired audio from a complex noisy environment, such as inmany ASR applications.

Further, under the present invention, audio beam 103 a may be generatedby an element with Cardioid beam, pattern, which has considerablesensitivity in all directions except the desired audio direction inspace. Resultantly, unlike the prior arts where there are some “blindspot” directions with incomplete noise cancellation, the presentinvention may cancel noise coming from virtually any one of a number ofdirections.

Acoustic Device

As alluded to earlier, acoustic device 102 may be formed compactly usingas few as two acoustic elements. FIGS. 2a-2g illustrate a number ofthese embodiments, from the perspective of the audio beams formed.However, the present invention is not so limited. In alternateembodiments, two or more acoustic elements may be used to responsivelygenerate beams 103 a and 103 b instead. For ease of understanding, thedescription will primarily be presented in the context of the varioustwo elements embodiments. Moreover, acoustic device 102 will simply bereferred to as “microphone”.

FIGS. 2a-2c illustrate three two-element embodiments, where one element202 in response to the presence of audio generates a cardioid beam andthe other element 204, in response to the presence of audio generates anomni directional beam. In each of these embodiments, cardioid beamgenerating acoustic element (hereinafter, simply “mic”) 202, is arrangedwith the null of the cardioid mic facing the expected originatingdirection of desired audio.

For the embodiment of FIG. 2a , omni-directional beam generating mic 204is arranged to face the expected originating direction of desired audioin parallel with cardioid beam generating mic 202. For each of theembodiments of FIG. 2b-2c , omni-directional beam generating mic 204 isarranged to face the expected originating direction of desired audio inseries with cardioid beam generating mic 202. For the embodiment of FIG.2b , omni-directional beam generating mic 204 is arranged to be disposed“behind” cardioid beam generating mic 202. For the embodiment of FIG. 2c, omni-directional beam generating mic 204 is arranged to be disposed“in front of” cardioid beam generating mic 202 (both viewed from theperspective of the expected originating direction of the desired audio).

FIG. 4a illustrates the corresponding acoustic beams 103 a and 103 b (inthe form of a “polar sensitivity” plot) responsively generated byelements 202 and 204 of the arrangements of FIGS. 2a-2c . As describedearlier, beam 103 a comprises a null facing in the originating directionof desired audio. For these embodiments, beam 103 b “radiates” in alldirections, and does not contain any null.

The null of cardioid beam generating element 202 is an attempt toeliminate the leakage of desired acoustic into beam 103 a. In reality,the null can often achieve as much as −20 dB attenuation relative to thesensitivity at the opposite direction. Nevertheless, experience hasshown that the present invention still exhibits consistent improvedresults over that of the prior arts.

Typically, the two acoustic elements are disposed proximally adjacent toeach other to enable the desired compact human interface be formed (forcertain applications). For these applications, the separating distancebetween two discrete mic elements may be in the range as small as 0.2 cmto 1 cm. For semiconductor acoustic devices, the separating distance maybe in the order of microns or even sub-microns. While care should beexercised to reduce the likelihood of cross interference between theelements, as illustrated by FIG. 2a-2c , their relative dispositions,i.e. whether facing the expected originating direction of desired audioin parallel or in series, are not as important as their respective beampatterns.

FIGS. 2d-2f illustrate three alternate two-element embodiments, whereboth elements are cardioid beam generating mics 202 a-202 b. In each ofthese embodiments, one of the two cardioid beam generating mics 202 a isarranged with its null facing the expected originating direction ofdesired audio, and the other cardioid beam generating mic 202 b arrangedwith its null facing away from the expected originating direction ofdesired audio.

For the embodiment of FIG. 2d , the other cardioid beam generating mic202 b is arranged to have its null face away from the expectedoriginating direction of desired audio, in parallel with the firstcardioid beam generating mic 202 a. Similarly, for each of theembodiments of FIG. 2e-2f , the other cardioid beam generating mic 202 bis also arranged with its null to face away from the expectedoriginating direction of desired audio, except in series with the firstcardioid beam generating mic 202 a.

For the embodiment of FIG. 2e , the other cardioid beam generating mic202 b is arranged to be disposed “behind” the first cardioid beamgenerating mic 202 a, whereas for the embodiment of FIG. 2f , the othercardioid beam generating mic 202 b is arranged to be disposed “in frontof” the first cardioid beam generating mic 202 a (both viewed from theperspective of the expected originating direction of the desired audio).

FIG. 4b illustrates the corresponding acoustic beams 103 a and 103 b(also in the form of a “polar sensitivity” plot) responsive generated byelements 202 a and 202 b of the arrangements of FIGS. 2d-2f . Asdescribed earlier, beam 103 a comprises a null facing in the originatingdirection of desired audio. For these embodiments, beam 103 b comprisesa null facing away from the originating direction of desired audio.

FIG. 2g illustrates yet another alternate two-element embodiment ofacoustic device 102. For this embodiment, two omni-directional beamgenerating mics 204 a and 204 b are used instead. The two elements 204 aand 204 b are arranged to face the expected originating direction ofdesired audio in series. The arrangement is supplemented with thecircuitry of FIG. 3 comprising delay 312, amplifier 314 and adder 316,implementing a “delay and sum” beam forming method.

As before, the responsive output of the second omni beam generating mic204 b provides beam 103 b. However, beam 103 a is formed by having adelay added to the responsive output of the first omni beam generatingmic 204 a, using delay 312, amplified using amplifier 314, and thensubtracted from beam 103 b.

The delay should be chosen so that the cardioid null is sufficientlydeep across all frequency in the bandwidth. The two acoustic elementsmay be balanced by adjusting the gain of amplifier 314, to avoidmismatch and reduction of the null.

The circuitry of FIG. 3 may be integrally disposed as part of acousticdevice 102, or it may be integrally disposed as part of signalprocessing subsystem 104. In yet other embodiments, the role of the twoomni beam generating mic 204 a and 204 b may be reversed.

Additionally, in addition to the “no null” and “single face away null”shape of FIGS. 4a and 4b , beam 103 b may comprise two or more nulls, aslong as none of the nulls is facing the originating direction of desiredaudio.

For example, FIG. 4c illustrates an alternate “clover leaf” beam shape(in a “polar sensitivity” plot) for beam 103 b having two “leafs”,forming two nulls, with the two nulls facing two directions 406 a and406 b that are substantially orthogonal to the originating direction ofdesired audio. FIG. 4d illustrates yet another alternate “clover leaf”beam shape (in a “polar sensitivity” plot) for beam 103 b having alsotwo “leafs”, forming two nulls, with the two nulls facing two directions406 c and 406 d, each forming an obtuse angle with the originatingdirection of desired audio.

In summary, acoustic device 102 comprises two or more acoustic elementsdesigned and arranged in a manner that facilitates generation of twosignals with one signal comprising mostly undesired audio, substantiallyvoid of desired audio, and another signal comprising both desired andundesired audio. The two or more acoustic elements may e.g. respond tothe presence of audio, desired and undesired, outputting a cardioid beamhaving a null facing the originating direction of desired audio, andanother beam having any one of a number of complementary beam shapes (aslong as it does not comprise a null facing the originating direction ofdesired audio).

Signal Processing Subsystem

FIG. 5 illustrates signal processing subsystem of FIG. 1 in furtherdetail, in accordance with one embodiment. As illustrated, for theembodiment, signal processing subsystem 104 comprises two channels ofinput (labeled as “reference” and “primary”), sampling components 502,optional pre-extraction signal conditioning components 504, signalextraction component 506, and optional post-extraction signalconditioning component 508. The elements are coupled to each other asshown.

Reference channel is employed to receive beam 103 a, whereas primarychannel is employed to receive beam 103 b.

Sampling components 504 are employed to digitized beams 103 a and 103 b.Typically, they are both digitized synchronically at the same samplingfrequency, which is application dependent, and chosen according to thesystem bandwidth. In the case of ASR applications, the samplingfrequency e.g. may be 8 kHz, 11 kHz, 12 kHz, or 16 kHz.

Typically, optional pre-extraction and post-extraction signalconditioning components 504 and 508 are application and/or extractiontechnique dependent. For example, in the case of ASR applications, andcertain signal extraction techniques, such as echo cancellation likeNLMS processing, pre-extraction and post-extraction signal conditioningcomponents 504 and 508 may be pre-whitening and de-whitening filters.The pre-whitening and de-whitening filters are employed to level, andreverse the leveling of the spectrum density of both signals. Levelingof the spectrum density of both channels improve NLMS converging speed,for uneven frequency distribution of the signals. Other single channelnoise cancellation technique, such as spectrum subtraction, can be addedas additional stage of optional post extraction signal conditioningcomponents.

Sampling Component

FIGS. 6a-6b illustrate the sampling component of FIG. 5 in furtherdetails, in accordance with two embodiments. For the embodiment of FIG.6a , sampling component 502 comprises two A/D converters 606, one eachfor the two beams 103 a and 103 b. Further sampling component 502includes pre-amps 602 and anti-aliasing filters 604. The elements arecoupled to each other as shown.

The signal from each acoustic element is amplified by a correspondingpre-amp 602, and then band-limited by a corresponding anti-aliasingfilter 604, before being digitized by a corresponding A/D converter atthe sampling frequency Fs.

FIG. 6b illustrates an alternate embodiment, where only one A/Dconverter 606 is used. However, sampling component 502 further includessample and hold components 608 and multiplexor 610. The elements arecoupled to each other as shown.

Each signal goes through the same processing as in FIG. 6a until afteranti-aliasing filtering (using anti-aliasing filters 604), then it issampled by sample-and-hold (S/H) unit 608 to produce a discreet signal.The output is then multiplexed (using multiplexor 610) with the discreetsignal from the other channel. Finally, the multiplexed signal isdigitized by A/D converter 606 into digital signal at twice the samplingfrequency (2×Fs).

Pre-Whitening and De-Whitening

As described earlier, for certain acoustic applications, such as ASRapplications, which tend to have stronger lower frequency componentsthan higher frequency components, it may be desirable to performpre-extraction conditioning of the signals, such as spectrum densityleveling through pre-whitening filtering, and therefore, post-extractionreverse conditioning, such as reversing the spectrum density levelingthrough de-whitening filtering.

For these applications, a pre-whitening filter (also referred to asde-colorization filter) is placed on both the primary and referenceinputs before they are sent to signal extraction component 506, inparticular, if component 506 implements NMLS noise cancellationprocessing, to alleviate the potential slow convergence rate broughtabout by narrow band (highly auto-correlated) input signal.

One embodiment each of a pre-whitening filter, and a de-whitening filteris illustrated in FIGS. 7a and 7b respectively.

For the embodiment of FIG. 7a , pre-whitening filter 504 is in the formof a pre-emphasis filter characterized by the equation:y _(n) =x _(n) −α*x _(n-1)

For the illustrated implementation, pre-whitening filter 504 includesstorage elements 702 and 704 for storing the preceding input value,x_(n-1), and the constant α, and multiplier 706 as well as adder 708.The elements are coupled to each other as shown, and collectivelyoperate to implement the processing to compute output y_(n), per theabove equation.

In alternate embodiment, pre-whitening filter 504 may also beimplemented in software.

FIG. 7b illustrates the complementary de-whitening filter, in the formof a de-emphasis filter, characterized by the equation:y _(n) =x _(n) +α*y _(n-1)

For the illustrated implementation, de-whitening filter 508 includesstorage elements 722 and 724 for storing the preceding output value,y_(n-1), and the constant α, and multiplier 726 as well as adder 728.The elements are coupled to each other as shown, and collectivelyoperate to implement the processing to compute output y_(n), per theabove equation.

Similarly in alternate embodiment, de-whitening filter 508 may also beimplemented in software.

Signal Extraction Component

FIG. 8 illustrates signal extraction component of FIG. 5 in furtherdetail, in accordance with one embodiment. The embodiment implements anecho cancellation like technique to recover desired audio by removingthe reference signal from the primary channel. The technique is referredto as “echo cancellation” like because similar to conventional “echocancellation”, one signal is subtracted from another. However, inclassical “echo cancellation”, the original signal that generates “echo”is accessible; and that original signal is not corrupted with thedesired audio. Whereas under the present invention, the original noisesignals are not available. Although the reference signal in the currentinvention is “substantially void of desired audio”, it still containssome desired audio. Extra steps, such as inhibition, should be taken toavoid the cancellation of the desired signal. In classical “echocancellation”, the “echo” signal that is being subtracted from thecomposite signal with the desired audio and echo, and more importantly,the “echo” signal bears a natural deterministic time lagged relationshipto the original audio that generated the echo. In contrast, under thepresent invention, it is the filtered reference signal, substantiallyvoid of the desired audio, being subtracted from the signal with bothdesired and undesired audio, and the reference and the desired signalsare acquired virtually concurrently responsive to the presence ofdesired and undesired audio.

Accordingly, in addition to echo cancellation like logic 810, signalextraction component 506 includes in particular a delay element 802 toartificially introduce a deterministic delay to the signal formed basedon beam 103 b (i.e. the signal on the primary channel). Thisartificially introduced delay enables modeling of reverberation betweenthe acoustic elements of acoustic device 102. Further, it enablesadaptive FIR filters employed in the echo cancellation like signalprocessing technique to approximate a non-causal filter.

The amount of delay to be artificially introduced in order to modelreverberation is application dependent. In general, it is approximatelyin the order of the duration of the impulse response of the environment,the delay then corresponds to an adaptive filter length which isapproximately twice the impulse response of the environment in order tomodel reverberation; the delay allows the adaptive filter to model thereverberation history. In various applications, the amount ranges from30 ms-60 ms for automotive environment, and 100 ms-200 ms for an officeenvironment.

For the embodiment, the echo cancellation like extraction of desiredaudio is actually conditionally operated, only when the channels areconsidered to be both active. Thus, beside signal extraction logic 810and delay element 802, for the embodiment, signal extraction component506 further includes mean amplitude estimation components 804, channelsignal detectors 806 and inhibition logic 808. Channel signal detectors806 are also collectively referred to as the “comparator” component, andin one embodiment, include in particular, two channel-active detectors,one each for the reference channel and the primary channel, and adesired audio detector. The elements are coupled to each other, and tothe earlier enumerated elements as shown.

Mean amplitude estimator components 804 are employed todetermine/estimate the power or amplitude of the signals of bothchannels for channel signal detectors 806, i.e. channel-active detectorand desired audio detector, as well as for the echo cancellation likesignal extraction process.

In one embodiment, the echo cancellation like signal extraction processimplemented is an adaptive noise cancellation process employing a NLMSFIR filter (FIG. 11a ). In other embodiments, the echo cancellation likesignal extraction processes implemented are adaptive noise cancellationprocesses employing a number of frequency domain LMS filters (FIG.11b-11c ). In yet other embodiments, the echo cancellation like signalextraction processes implemented are adaptive noise cancellationprocesses employing a number subband LMS filter (FIG. 11d-11e ).

These elements are further described in turn below.

Mean Amplitude Estimator

FIG. 9 illustrates the mean amplitude estimation component of FIG. 8 infurther detail, in accordance with one embodiment. For the embodiment,mean amplitude estimation component 804 calculates a weighted runningaverage of the absolute value of the input as characterized by theequation:y _(n)=(1−α)*y _(n-1) +α*|x _(n)|

The weight coefficient determines the length of the running window.

The embodiment includes various storage elements 902-908 for storing thevalues of |x_(n)|, y_(n-1), α, and (1−α) respectively, and multipliers910 and adder 912 to perform the computations.

As with the earlier described pre-whitening and de-whitening components.mean amplitude estimation component 804 may also be implemented insoftware.

Detectors

FIGS. 10a-10b illustrate the comparator component, i.e. detectors, ofFIG. 8 in further detail, in accordance with one embodiment. Morespecifically, FIG. 10a shows the logic of a desired audio detector 806a, whereas FIG. 10b shows the logic of a channel active detector 806 b.

For the illustrated embodiment, desired audio detector 806 a includesstorage element 1002 for storing an audio threshold offset.Additionally, desired audio detector 806 a further includes ratiocalculator 1004, long term running mean amplitude ratio value calculator1006, adder 1008 and comparator 1010. The elements are coupled to eachother as shown. The embodiment is a power based detector.

Ratio calculator 1004 is employed to calculate the ratio of the primaryand reference signal mean amplitude. Running mean amplitude ratiocalculator 1006 is employed to calculate the long term running meanvalue of the ratio, which provides the base or floor for the desiredaudio. Comparator 1010 and adder 1008 are employed to compare thecurrent ratio to determine whether it is greater than the long termrunning ratio by at least a threshold offset. If it is above the base byat least the threshold offset, desired audio is considered detected;otherwise no desired audio is assumed.

The embodiment is designed for desired audio that tends to exhibit abursty characteristic, such as speech. For other audio applications,suitably modified embodiments may be employed instead.

In alternate embodiments, other desired audio detectors, e.g.correlation based desired signal detector, may be employed instead. FIG.10b shows a channel-active detector in further detail, in accordancewith one embodiment. The embodiment is a power based comparator. Asillustrated, channel-active detector 806 b comprises storage element1024 for storing a threshold value, and comparator 1026 for comparingthe mean amplitude of the channel to the stored threshold value. If it'sabove the stored threshold value, the channel is assumed to be active;otherwise the channel is assumed to be inactive.

Further, as with the earlier described pre-whitening/de-whitening andmean amplitude estimation components, detectors 804 a-804 b may also beimplemented in software.

Inhibit

FIG. 12 illustrates the operating logic of the inhibit component of FIG.8 in further details, in accordance with one embodiment. As illustrated,for the embodiment, designed for time domain implementation, inhibitcomponent 808 using the inputs provided detectors 806 first determineswhether both the primary and the reference channels are active, blocks1202-1204. If either the primary or the reference channel is determinedto be inactive, the inhibit signal is set to “positive”, resulting insubstantial inoperation of the signal extraction block (e.g. to conservecomputing power), i.e. no signal extraction (filtering) or adjustment tothe extraction (adaptation) is performed. Under the condition, for theembodiment, whatever signal is present on the primary channel isoutputted, block 1210.

However, if both channels are active, inhibit logic 808 furtherdetermines if either desired audio is present or a pause threshold (alsoreferred to as hangover time) has not been reached, block 1206-1208. Thepause threshold (or hangover time) is application dependent. Forexample, in the case of ASR, the pause threshold may be a fraction of asecond.

If the desired audio is detected or the pause time is not exceeded, theinhibit signal is set to “positive with filter adaptation disabled”,i.e. filtering coefficients frozen, block 1212. The reference signal isfiltered accordingly, and subtracted from the primary channel togenerate desired audio.

If the desired audio is not detected, and the pause time is exceeded(but the channels are active), the inhibit signal is set to “negativewith filter adaptation enabled”, block 1214. Under the condition, thefiltering coefficients of the employed filters will be adapted.

Note that the above described embodiment advantageously employs thedetectors and inhibition before the primary signal (having the desiredaudio) delay, thus the likelihood of the desired audio negativelyimpacting the filter adaptation operation is reduced.

As alluded to earlier, described in more detail below, filtering mayalso be performed in the frequency and subband domains. For theseembodiments, the above inhibit implementation may be practiced on afrequency by frequency or subband by subband basis.

Echo Cancellation Like Signal Extraction Component

FIGS. 11a-11e illustrate echo cancellation like signal extractioncomponent of FIG. 8 in further detail, in accordance with variousembodiments. More specifically, FIG. 11a illustrates an implementationemploying a NLMS adapted approach, whereas FIGS. 11b-11c illustrate twoimplementations employing a frequency domain LMS adapted approach. FIG.11d-11e illustrate two implementations employing a subband LMS adaptedapproach.

As illustrated in FIG. 11a , echo cancellation like signal extractioncomponent 810 of the NLMS adaptive implementation comprises adaptive FIRfilter 1102 and adder 1104. The elements are coupled to each other asshown.

The (conditioned) signal of the reference channel is filtered byadaptive FIR filter 1102, and subtracted from the delayed signal of theprimary channel, using adder 1104. The result is outputted as thedesired audio.

The extraction logic operates as a loop running on a sample-by-samplebasis. The reference signal is filtered by the adaptive FIR filter 1102.Essentially, a transfer function is applied to the reference channel tomodel the acoustic path from the cardioid element to the other element,so that the filtered reference signal closely matches the noisecomponent of the signal in the primary channel. The filtered referencesignal is then subtracted from the delayed primary signal. What is left,is the desired audio.

The output of the NLMS is also called the NLMS error; it is used toadjust the adaptive FIR filter coefficients so that the NLMS error willbe minimized when the desired audio is not present.

As illustrated in FIG. 11b , echo cancellation like signal extractioncomponent 810 of the first frequency LMS implementation comprises FFTcomponents 1112, a number of adaptive filters 1114 (two shown), a numberof adders 1116 (two shown) and IFFT component 1118. The elements arecoupled to each other as shown.

The (conditioned) signals of the reference channel and the delayedprimary channel are first “decomposed” into a number of frequencycomponents (two shown), by the corresponding FFT components 1112. Eachof the frequency components of the reference signal is filtered by acorresponding adaptive filter 1114, and subtracted from thecorresponding frequency component of the delayed signal of the primarychannel. using a corresponding adder 1116. The resulted frequencycomponents are “recombined”, using IFFT component 1118, and therecombined signal is outputted as the desired audio.

As illustrated in FIG. 11c , echo cancellation like signal extractioncomponent 810 of the second frequency LMS implementation comprises FFTcomponents 1122 a-1122 b, a number of adaptive filters 1124 (two shown),adder 1128 and IFFT component 1126. The elements are coupled to eachother as shown.

The (conditioned) signal of the reference channel is first “decomposed”into a number of frequency components (two shown), by FFT component 1122a. Each of the frequency components of the reference signal is filteredby a corresponding adaptive filter 1124. The filtered frequencycomponents are recombined into a filtered reference signal, using IFFTcomponent 1126, which is then subtracted from the delayed signal of theprimary channel, using adder 1128, to generate desired audio.

The error signal (comprising the desired audio, if present) is also“decomposed” into a number of frequency components, using FFT component1122 b, and the “decomposed” frequency components are used to adaptfilters 1124.

As illustrated in FIG. 11d , echo cancellation like signal extractioncomponent 810 of the first subband LMS implementation comprises analysisbanks 1132 a-1132 b, a number of down-sampling units 1134 a-1134 b (twosets of two shown), a number of adaptive filters 1136 (two shown), anumber of adders 1138 (two shown), a number of up-sampling units 1140(two shown), and a synthesis bank 1142. The elements are coupled to eachother as shown.

The (conditioned) signals of the reference channel and the delayedprimary channel are first “decomposed” into a number of subbandcomponents (two shown), by the corresponding analysis banks 1132 a/1132b. Each of the subband components of the reference signal is firstdown-sampled by a predetermined factor, using a correspondingdown-sampling unit 1134, and then filtered by a corresponding adaptivefilter 1136. Each of the filtered subband components is then subtractedfrom the corresponding subband component of the delayed signal of theprimary channel, using a corresponding adder 1138. The resulted subbandcomponents are up-sampled by the same factor, using a correspondingup-sampling unit 1140, and then “recombined”, using synthesis bank 1142.The recombined signal is outputted as the desired audio.

As illustrated in FIG. 11e , echo cancellation like signal extractioncomponent 810 of the second subband LMS implementation comprisesanalysis banks 1152 a-1152 b, a number of down-sampling units 1154 a and1154 b (two sets of two shown), a number of adaptive filters 1156 (twoshown), a number of up-sampling units 1158 (two shown), synthesis bank1160 and adder 1162. The elements are coupled to each other as shown.

The (conditioned) signal of the reference channel is first “decomposed”into a number of subband components (two shown), by analysis bank 1152a. Each of the subband components of the reference signal is downsampled by a predetermined factor, using a corresponding down samplingunit 1154 a, and then filtered by a corresponding adaptive filter 1156.The filtered subband components are up-sampled by correspondingup-sampling units 1158, and then recombined into a filtered referencesignal, using synthesis bank 1160, which is then subtracted from thedelayed signal of the primary channel, using adder 1162, to generatedesired audio.

The error signal (comprising the desired audio, if present) is also“decomposed” into a number of subband components, using analysis bank1152 b and down sampling unit 1154 b, and the “decomposed” subbandcomponents are used to adapt filters 1156.

Each of these signal extraction component embodiments may also beimplemented in software.

Blind Signal Separation

FIG. 13 illustrates signal extraction component 506 of FIG. 5 in furtherdetail, in accordance with another embodiment. As opposed to the earlierdescribed echo cancellation like embodiments, signal extractioncomponent 506 of FIG. 13 implements a blind signal separation techniqueto remove the signal of the reference channel from the signal of theprimary channel to extract desired audio.

As illustrated, signal extraction component 810 comprises a number ofadaptive FIR filters 1302, adders 1306 and a cost function 1304. Theelements are coupled to each other as shown.

Both the reference and the primary channels are filtered using adaptiveFIR filters 1302. The results are subtracted from each other usingcorresponding adders 1306. The resulted signals are outputted, with theresult of the reference signal having been subtracted from the primarysignal being the desired audio.

The output signals are in turn feedback to a cost function, whichoutputs respective adaptation for adaptive FIR filters 1302 based on thetwo output signals. The cost function depends on specific BSS method.

CONCLUSION AND EPILOGUE

Thus, it can be seen from the above descriptions, various novel acousticdevices, systems and methods have been described.

While the present invention has been described in terms of the abovedescribed embodiments, those skilled in the art will recognize that theinvention is not limited to the embodiments described. The presentinvention can be practiced with modification and alteration within thespirit and scope of the appended claims. Thus, the description is to beregarded as illustrative instead of restrictive on the presentinvention.

What is claimed is:
 1. A device to detect desired audio, comprising: aratio calculator, the ratio calculator calculates a ratio between aprimary acoustic signal, and a reference acoustic signal, the primaryacoustic signal contains desired audio and undesired audio and thereference acoustic signal contains mostly undesired audio, substantiallyvoid of desired audio: a long-term mean value calculator, the long-termmean value calculator is coupled to the ratio calculator, the long-termmean value calculator maintains a running average of the ratio; and acomparator, the comparator is coupled to the ratio calculator and thelong-term value calculator, the comparator compares the ratio with therunning average, desired audio is detected when the ratio is greaterthan the running average by a threshold amount.
 2. The device of claim1, further comprising: an adder, the adder is coupled to the long-termmean value calculator and to the comparator, the adder adds a thresholdvalue to the running average.
 3. The device of claim 1, furthercomprising: logic, the logic enables the reference acoustic signal to befiltered when desired audio is detected, filtering is performed with anadaptive FIR filter to obtain a filtered reference acoustic signal andthe filtered reference acoustic signal is subtracted from a delayedprimary acoustic signal to obtain desired audio.
 4. The device of claim3, wherein the reference acoustic signal is filtered, when a pause timeis not exceeded, with the adaptive FIR filter to obtain the filteredreference acoustic signal; and the filtered reference acoustic signal issubtracted from a delayed primary acoustic signal to obtain desiredaudio.
 5. The device of claim 3, wherein when desired audio is notdetected, and a pause time is exceeded and the primary channel is activeand the reference channel is active, coefficients of the adaptive FIRfilter are adapted.
 6. The device of claim 3, wherein a delay applied tothe primary acoustic signal is approximately equal to an impulseresponse time of an environment and a length of the adaptive FIR filteris equal to twice the delay.
 7. The device of claim 6, wherein amagnitude of the delay is in the range of approximately 30 millisecondsto 200 milliseconds.
 8. A method to detect desired audio, comprising:creating an amplitude ratio between a primary acoustic signal and areference acoustic signal, wherein the primary acoustic signal containsdesired audio and undesired audio and the reference acoustic signalcontains mostly undesired audio, substantially void of desired audio;averaging the amplitude ratio over time to provide a floor for desiredaudio; and comparing the amplitude ratio to the floor, wherein: (a)desired audio is detected when the amplitude ratio exceeds the floor byan offset and (b) desired audio is not detected when the amplitude ratiodoes not exceed the floor by the offset.
 9. The method of claim 8,further comprising: filtering the reference acoustic signal when desiredaudio is detected, the filtering is performed with an adaptive FIRfilter to obtain a filtered reference acoustic signal; and subtractingthe filtered reference acoustic signal from a delayed primary acousticsignal to obtain desired audio.
 10. The method of claim 8, furthercomprising: filtering the reference acoustic signal when a pause time isnot exceeded with an adaptive FIR filter to obtain a filtered referenceacoustic signal and subtracting the filtered reference acoustic signalfrom a delayed primary acoustic signal to obtain desired audio.
 11. Themethod of claim 8, when desired audio is not detected and a pause timeis exceeded and the primary channels active and the reference channel isactive, the method further comprising: adapting coefficients of theadaptive FIR filter.
 12. The method of claim 10, wherein in an automaticspeech recognition application the pause time is approximately afraction of a second.
 13. The method of claim 9, wherein a delay appliedto the primary acoustic signal is approximately equal to an impulseresponse time of an environment and a length of the adaptive FIR filteris equal to twice the delay.
 14. The method of claim 13, wherein amagnitude of the delay is in the range of approximately 30 millisecondsto 200 milliseconds.
 15. An apparatus comprising: an acoustic signalprocessing system, the acoustic signal processing system is configuredto receive a primary channel and a reference channel; a computerreadable medium containing executable computer program instructions,which when executed by the acoustic signal processing system. cause theacoustic signal processing system to perform a method comprising:creating an amplitude ratio between a primary acoustic signal and areference acoustic signal. wherein the primary acoustic signal containsdesired audio and undesired audio and the reference acoustic signalcontains mostly undesired audio, substantially void of desired audio;averaging the amplitude ratio over time to provide a floor for thedesired audio; and comparing the amplitude ratio with the floor,wherein: (a) desired audio is detected when the amplitude ratio exceedsthe floor by an offset, and (b) desired audio is not detected when theamplitude ratio does not exceed the floor by the offset.
 16. The apprates of claim 15, wherein the method further comprises: filtering thereference acoustic signal when desired audio is detected, the filteringis performed with an adaptive FIR filter to obtain a filtered referenceacoustic signal; and subtracting the filtered reference acoustic signalfrom a delayed primary acoustic signal to obtain desired audio.
 17. Theapparatus of claim 15, wherein the method further comprises: filteringthe reference acoustic signal when a pause time is not exceeded with anadaptive FIR filter to obtain a filtered reference acoustic signal; andsubtracting the filtered reference acoustic signal from a delayedprimary acoustic signal to obtain the desired audio.
 18. The apparatusof claim 15, when desired audio is not detected and a pause time isexceeded and the primary channel is active and the reference channel isactive, the method further comprises: adapting coefficients of theadaptive FIR filter.
 19. The apparatus of claim 17, wherein in anautomatic speech recognition application the pause time is approximatelya fraction of a second.
 20. The apparatus of claim 16, wherein a delayapplied to the primary acoustic signal is approximately equal to animpulse response time of an environment and a length of the adaptive FIRfilter is equal to twice the delay.